US7756052B2 - Determining quality of voice calls over a packet network - Google Patents
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1069—Session establishment or de-establishment
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L67/00—Network arrangements or protocols for supporting network services or applications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
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- the present invention relates to internet telephony techniques, and more particular, to a method for determining voice quality in a voice call over a packet network and rerouting the voice call accordingly.
- the objective methods for measuring voice quality of a voice call over a packet network such as the Internet using a protocol such as VOIP or SIP are mainly categorized into three types: 1) intrusive method: in this method, a reference signal is injected on one side of the network and collected on the other. An assessment of the quality of the network can be made by comparing any differences between the original and transmitted signals. This method gives a true end-to-end quality assessment, however, it is intrusive, requires a receiver to collect the transmitted signal, and does not account for network latency; 2) signal-based non-intrusive method: in this method, the voice quality is assessed by “listening” to ordinary phone calls and applying complex speech pattern recognition.
- This method is non-intrusive, and gives true end-to-end measurements. However, it does not account for network delay either. Moreover, it is impractical for a large-size fully-meshed network; and 3) parameter-based non-intrusive method: this method assesses voice quality by collecting impairment information from ordinary phone calls on the packet network. It uses the known E-model formula to derive the quality of a call. This method does not give true end-to-end results but only gives the voice quality of the packet network portion of the phone call.
- the present invention is directed to the method of the third type as explained above, i.e., the parameter-based non-intrusive method.
- a method for managing a voice call over a packet switched network in which the voice call is rerouted when voice quality is below a predetermined level.
- the voice quality is determined by calculating a parameter representing the voice quality based on information regarding a codec, network delay and packet loss.
- the information regarding codec, network delay and packet loss is obtained from one or more gateways.
- the gateways comprise at least one Cisco gateway, and the information is obtained from Call Detail Records available in the Cisco gateway.
- the R-factor is determined based on the information regarding codec, network delay and packet loss from the following equations, wherein d cdr and the PL are the network delay and the packet loss provided by Cisco CDR, and:
- the R-factor is determined from the following equations:
- FIG. 1 schematically illustrates a voice call over a packet switched network using VOIP protocol
- FIG. 2 is a flow chart showing the embodiment of the method according to the present invention.
- FIG. 1 schematically illustrates a scheme of a typical telephone call which is arranged over a packet network using a proper protocol such as VOIP.
- a call initiated from a calling telephone 1 transmits over a link in a PSTN network 6 .
- the call arrives at an originating gateway 2 , where the PSTN call is translated into packet data and routed through a internet 3 to a terminating gateway 4 , where the packet data is translated back into a PSTN call forwarded to a destination telephone 5 . Therefore, there are two PSTN legs and one VOIP leg to complete the call.
- Mean Opinion Score is a standard published by the ITU to evaluate a voice call.
- the ITU has published other standards to objectively measure call quality, including a non-intrusive, parameter-based objective method called “E-model”.
- E-model a non-intrusive, parameter-based objective method
- the preferred embodiments of the present invention are described with E-model applied to calculate the MOS score to measure the quality for a call through a VOIP network.
- the MOS can be obtained from R-factor according to Table 1:
- the R-factor is preferably calculated based on information regarding the codec, network delay and packet loss obtained from the network, preferably from one or more gateways in the network.
- the codec, network and packet loss information is obtained from Call Detail Records (CDRs) available in one or more Cisco gateways in the network.
- CDRs Call Detail Records
- the information is obtained during the call session.
- the R-factor and the MOS score are calculated immediately with the obtained information so as to dynamically monitor the call quality during the call.
- the calculation results are fed back to a management unit for back-end processing, and are used to update routing tables.
- a routing engine immediately reroutes the call according to the updated routing tables.
- the call can be rerouted in real time during the call session.
- FIG. 2 schematically illustrates the steps of a method for managing a voice call over a VOIP network according to a preferred embodiment of the present invention.
- the call starts at block 100 .
- information regarding the codec, network delay and packet loss is obtained from the network during the call, at block 101 .
- information regarding the codec, network delay and packet loss is obtained from the network during the call, at block 101 .
- information is obtained from Call Detail Records (CDRs) available in one or more Cisco gateways.
- CDRs Call Detail Records
- Cisco gateway can either be an originating gateway or a terminating gateway, or both.
- R-factor is then calculated based on the obtained information regarding the codec, network delay and packet loss according to a proper formula (examples given in detail below), at block 102 .
- MOS is calculated from R-factor according the Equation 1 or Table 1, as explained above.
- the calculation results are compared with a predetermined reference value which represents an acceptable voice quality threshold level, at block 103 . If R or the MOS is above the threshold, the call is kept in the path without being rerouted, and periodically the steps 101 - 103 are repeated so as to monitor the call quality. If R or the MOS falls below the threshold, as decided in block 103 , the routing table is updated and the call is rerouted, at block 105 .
- steps 101 - 104 are repeated to keep monitoring the call, until the call ends at block 105 .
- the Expectancy Factor A is used to increase the MOS score for destinations with inherent poor quality such as cell phones, hard to reach areas and developing countries. Since customers have come to expect lower quality to these destinations, what might otherwise sound poorly to a “normal” destination is acceptable in these cases. Therefore, in this embodiment according to the present invention, is assumed to be zero “0”.
- R o ⁇ I s are constants throughout the conversation in the call session, regardless of the transmission medium.
- the impairment factor I d is a function of the mouth to ear one-way delay d.
- d n can be obtained from Cisco CDRs, the challenge becomes to derive default values for d c & d j that are gateway specific.
- d n In the lab we set d n to 0 and assumed d c and d j to be a combined constant that does not depend on network conditions.
- Equations 20, 21 and 22 are derived from curves generated from the G.113 I e data as listed in Table 2:
- G.113 I e data G.729 G.723 G.711 G.711 % PL (VAD) (VAD) w/o PLC w/PLC 0 11 15 0 0 0.5 13 17 1 15 19 25 5 1.5 17 22 2 19 24 35 7 3 23 27 45 10 4 26 32 5 55 15 7 20 8 36 41 10 25 15 35 16 49 55 20 45
- Equation 8 for calculating R:
- Expectancy Factor A is still assumed as “0” in this second embodiment.
- R o ⁇ I s 100 (Constant 1—revised)
- I d I dte +I dle +I dd (Equation 24)
- dc+dj is equal to 94 ms (for G.711) or 114 ms (for G.729) or 186 ms (for G.723) (see Constants 2, 3 and 4), we can therefore assume d to always be greater than 100 ms (even for G.711 codec, where the native codec delay is 94 ms, we can safely assume that network delay d cdr will be greater than 12 ms) and Equation 25 applies for all codecs and network conditions.
- Equation 8 for calculating R:
- the information regarding the codec, network delay and packet loss can be periodically and/or dynamically obtained from the network.
- the information may be obtained from other types of gateways or devices in the network.
- the present invention may be applied to voice calls over packet network using other protocols as well, such as SIP protocol. Therefore, the scope of the present invention is solely intended to be defined by the accompanying claims.
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Abstract
Description
MOS=1+0.035R+R(R−60)(100−R)7*10−6 (Equation 1)
R=93.2−0.024d−40 ln(1+10PL) if d=<177.3 (Equation 2a)
R=112.7−0.134d−40 ln(1+10PL) if d>177.3 (Equation 2b)
and d=d cdr/2+94
R=82.2−0.024d−44 ln(1+9PL) if d=<177.3 (Equation 3a)
R=101.7−0.134d−44 ln(1+9PL) if d>177.3 (Equation 3b)
and d=d cdr/2+114
R=97.7−0.134d−50 ln(1+8PL) (Equation 4)
and d=d cdr/2+186
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[0+40 ln(1+10PL)] (Equation 5)
where X=log(d/100)/log 2 and d=d cdr/2+94
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[11+44 ln(1+9PL)] (Equation 6)
where X=log(d/100)/log 2 and d=d cdr/2+114
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[15+50 ln(1+8PL)] (Equation 7)
where X=log(d/100)/log 2 and d=d cdr/2+186
MOS=1+0.035R+R(R−60)(100−R)7*10−6 (Equation 1)
TABLE 1 |
R-factor and MOS score comparison |
R-Factor | Perceive Quality | MOS | ||
90-100 | Best | 4.34-4.5 | ||
80-90 | High | 4.03-4.34 | ||
70-80 | Medium | 3.60-4.03 | ||
60-70 | Low | 3.10-3.60 | ||
<60 | Poor | <3.10 | ||
R=R o −I s −I d −I e +A (Equation 8)
-
- where: R: R-factor
- Ro: Basic Signal to noise ratio
- Is: Impairment occurring simultaneously with the voice signal
- Id: Impairment caused by one way delay
- Ie: A combined impairment caused by Codec & packet loss
- A: Expectancy Factor
- where: R: R-factor
R o −I s=93.2 (Constant 1)
I d=0.024d if d=<177.3 (Equation 9)
I d=0.024+0.11(d−177.3) if d>177.3 (Equation 10)
d=d c +d n +d j (Equation 11)
dc+dj=94 ms, for G.711 (Constant 2)
dc+dj=114 ms, for G.729 (Constant 3)
dc+dj=186 ms, , for G.723 (Constant 4)
where: d=d cdr/2+94 for G.711 codec (Equation 12)
d=d cdr/2+114 for G.729 codec (Equation 13)
d=d cdr/2+186 for G.723 codec (Equation 14)
d c +d j=15pl+94 for G.711 and PL<2% (Equation 15)
d c +d j=2pl+125 for G.711 and PL>=2% (Equation 16)
d c +d j=10pl+114 for G.729 and PL<2% (Equation 17)
d c +d j=1.25pl+132.5 for G.729 and PL>=2% (Equation 18)
d c +d j=40pl+186 for G.723 (Equation 19)
Ie=40 ln(1+10PL) for G.711 (Equation 20)
Ie=11+44 ln(1+9PL) for G.729 (Equation 21)
Ie=15+50 ln(1+8PL) for G.723 (Equation 22)
TABLE 2 |
G.113 Ie data |
G.729 | G.723 | G.711 | G.711 | |
% PL | (VAD) | (VAD) | w/o PLC | w/PLC |
0 | 11 | 15 | 0 | 0 |
0.5 | 13 | 17 | ||
1 | 15 | 19 | 25 | 5 |
1.5 | 17 | 22 | ||
2 | 19 | 24 | 35 | 7 |
3 | 23 | 27 | 45 | 10 |
4 | 26 | 32 | ||
5 | 55 | 15 | ||
7 | 20 | |||
8 | 36 | 41 | ||
10 | 25 | |||
15 | 35 | |||
16 | 49 | 55 | ||
20 | 45 | |||
R=93.2−0.024d−40 ln(1+10PL) if d=<177.3 (Equation 2a)
R=112.7−0.134d−40 ln(1+10PL) if d>177.3 (Equation 2b)
and d=d cdr/2+94
R=82.2−0.024d−44 ln(1+9PL) if d=<177.3 (Equation 3a)
R=101.7−0.134d−44 ln(1+9PL) if d>177.3 (Equation 3b)
and d=d cdr/2+114
R=97.7−0.134d−50 ln(1+8PL) (Equation 4)
and d=d cdr/2+186
R o −I s=100 (Constant 1—revised)
4.5=1+0.035R+R(R−60)(100−R)7×10−6 (Equation 23)
I d =I dte +I dle +I dd (Equation 24)
-
- where: Idte gives an estimate for the impairments due to talker echo
- Idle gives an estimate for the impairments due to listener echo
- Idd represents the impairment due to the end-to-end one way delay
- where: Idte gives an estimate for the impairments due to talker echo
Id=0 for d<100 ms
Id=25{(1+X 6)1/6−3(1+[X/3]6)1/6+2} for d>100 ms (Equation 25)
where X=log(d/100)/log 2
Ie=40 ln(1+10PL) for G.711 (Equation 20)
Ie=11+44 ln(1+9PL) for G.729 (Equation 21)
Ie=15+50 ln(1+8PL) for G.723 (Equation 22)
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[0+40 ln(1+10PL)] (Equation 5)
where X=log(d/100)/log 2 and d=d cdr/2+94
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[11+44 ln(1+9PL)] (Equation 6)
where X=log(d/100)/log 2 and d=d cdr/2+114
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[15+50 ln(1+8PL)] (Equation 7)
where X=log(d/100)/log 2 and d=d cdr/2+186
Claims (4)
R=93.2−0.024d−40 ln(1+10PL) if d=<177.3 (Equation 2a)
R=112.7−0.134d−40 ln(1+10PL) if d>177.3 (Equation 2b)
and d=d cdr/2+94
R=82.2−0.024d−44 ln(1+9PL) if d=<177.3 (Equation 3a)
R=101.7−0.134d−44 ln(1+9PL) if d>177.3 (Equation 3b)
and d=d cdr/2+114
R=97.7−0.134d−50 ln(1+8PL) (Equation 4)
and d=d cdr/2+186
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[0+40 ln(1+10PL)] (Equation 5)
where X=log(d/100)/log 2 and d=d cdr/2+94
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[11+44 ln(1+9PL)] (Equation 6)
where X=log(d/100)/log 2 and d=d cdr/2+114
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[15+50 ln(1+8PL)] (Equation 7)
where X=log(d/100)/log 2 and d=d cdr/2+186
R=93.2−0.024d−40 ln(1+10PL) if d=<177.3 (Equation 2a)
R=112.7−0.134d−40 ln(1+10PL) if d>177.3 (Equation 2b)
and d=d cdr/2+94
R=82.2−0.024d−44 ln(1+9PL) if d=<177.3 (Equation 3a)
R=101.7−0.134d−44 ln(1+9PL) if d>177.3 (Equation 3b)
and d=d cdr/2+114
R=97.7−0.134d−50 ln(1+8PL) (Equation 4)
and d=d cdr/2+186;
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[0+40 ln(1+10PL)] (Equation 5)
where X=log(d/100)/log 2 and d=d cdr/2+94
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[11+44 ln(1+9PL)] (Equation 6)
where X=log(d/100)/log 2 and d=d cdr/2+114
R=100−25{(1+X 6)1/6−3(1+[X/3]6)1/6+2}−[15+50 ln(1+8PL)] (Equation 7)
where X=log(d/100)/log 2 and d=d cdr/2+186
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US8477637B1 (en) * | 2006-12-31 | 2013-07-02 | At&T Intellectual Property Ii, L.P. | Method and apparatus for monitoring a packet network |
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US7706278B2 (en) * | 2007-01-24 | 2010-04-27 | Cisco Technology, Inc. | Triggering flow analysis at intermediary devices |
US8331358B2 (en) * | 2007-07-25 | 2012-12-11 | Actiontec Electronics, Inc. | Systems and methods for connecting a packet-based call to a conventional telephone network |
US9769237B2 (en) * | 2008-04-23 | 2017-09-19 | Vonage America Inc. | Method and apparatus for testing in a communication network |
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US9232048B2 (en) * | 2013-12-04 | 2016-01-05 | International Business Machines Corporation | Quality of experience determination for multi-party VoIP conference calls that account for focus degradation effects |
US20160112463A1 (en) * | 2014-10-16 | 2016-04-21 | Kanfield Capital Sa | Systems and methods for establishing a telephone connection |
CN106304180A (en) * | 2016-08-15 | 2017-01-04 | 中国联合网络通信集团有限公司 | A kind of method and device of the speech service quality determining user |
CN108091350A (en) * | 2016-11-22 | 2018-05-29 | 中国移动通信集团公司 | A kind of speech quality assessment method and device |
US10805191B2 (en) | 2018-12-14 | 2020-10-13 | At&T Intellectual Property I, L.P. | Systems and methods for analyzing performance silence packets |
US11611664B2 (en) * | 2021-03-25 | 2023-03-21 | Agora Lab, Inc | Voice quality assessment system |
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